similar to: Slightly OT: Getting VOIP number into phone book

HI All, I am using this code here: setTimeout("Sound.play(''sounds/movieprojector.mp3'');", 600); setTimeout("Effect.Appear(''slideshow5'');", 850 + 600); setTimeout("Effect.Puff(''slideshow5'');", 850 + 600 + 850);

Hi All, Has anyone here used zkoss.org? What do you think of it in comparison to scriptaculous? -- Leonard Burton, N9URK http://www.jiffyslides.com service-CbOvBfcOUrWrJCssh9Shfg@public.gmane.org leonardburton-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org "The prolonged evacuation would have dramatically affected the survivability of the occupants."

Greetings list, Wondering if anyone's come across this before. I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server: -- Called

Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there?

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Hi, Is it possible to dial voicemail from a particular phone line and automatically enter the extension that is being dialed from, thereby only prompting for the password? I've been searching around to find if this is possible, but I haven't been able to find an example of this. I have a feeling it's more of a endpoint function, but I thought I'd ask if anyone has accomplished

Hi, I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to make zaptel working... OS is gentoo linux 2006.1 Hardware: --------- 0000:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b400 Memory at ff900000

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Hi guys, Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? Alex

Greetings list, I posted this to the -biz list a few days ago. In hindsight, I think it would have been more appropriate posted here, so apologies to those on both lists who've now seen this twice. I have had a request to provide 2x PRIs to a site in Lausanne, Switzerland, but my knowledge of the Swiss Telco market is non-existent. Are there any folks on the list who've experience in

Greetings list, Not exclusively asterisk-related, but I've noticed the CentPBX site has been offline the last few days. Anyone know the reasoning behind that, and more importantly, is anyone mirroring it? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons

Greetings list, Wondering if some kind soul can help me with an issue with chan_sip segfaulting as soon as it loads... Basically, if sip.conf contains any peers with "host=dynamic" in them, asterisk won't start. Doing -vvvdddc yields the following: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Segmentation fault As

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Greetings all, Not specifically an asterisk query, but a couple of transfer queries that I'm sure are obvious to folks who use these phones all the time: 1) how does one do a blind transfer? When a call is answered and one hits the transfer button, followed by an extension, one has to wait for the other party to answer, then hit transfer again, before the call is released. I'm sure there

Don't know if anyone's got experiences on this they'd be able to share... I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a company selling them. There's one for Nigeria listed on didx.com, but $22/month seems a little steep. Has anyone had any luck getting DIDs for countries in Africa, and are there any companies out there selling them? Thanks

Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. This is a simple test number

Greetings list, Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc.. A cursory googling has led me to the Polycom Soundpoint IP4000 at around the ?450 mark - any thoughts on this? If anyone knows a good Polycom wholesaler in the UK, I'd be

Hi All, Until now I've only used IAX2 to connect to ITSPs. I've been toying with a SIP connection to Gizmo Project, but not yet successfully. It brings to mind a question. At what point does it make sense to consider a SIP-aware firewall such as those from Ingate? I'd hate to move away from my m0n0wall, which is open source, easy to manage and has served me brilliantly for two

On 19/10/2020 17:00, PGNet Dev wrote: > On 10/19/20 1:18 AM, John Fawcett wrote: >> I would recommend you to redo the tests after correcting the >> configuration. To be doubly sure you can include accented and unique non >> accented text in the same email and search for both. If the non accented >> text is found you know you've searching against the updated index and